- Moving forward declarations to one block
- Moving global variables to one block - Moving global networking variables to one block - Small whitespace changes - Renaming a few more global channel settings to global_ for clarity (No functional changes) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8825 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Olle Johansson
parent
7ac422562e
commit
4a63775523
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@ -140,8 +140,6 @@ static int expiry = DEFAULT_EXPIRY;
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#define CALLERID_UNKNOWN "Unknown"
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#define DEFAULT_MAXMS 2000 /*!< Qualification: Must be faster than 2 seconds by default */
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#define DEFAULT_FREQ_OK 60 * 1000 /*!< Qualification: How often to check for the host to be up */
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#define DEFAULT_FREQ_NOTOK 10 * 1000 /*!< Qualification: How often to check, if the host is down... */
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@ -152,12 +150,15 @@ static int expiry = DEFAULT_EXPIRY;
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#define SIP_MAX_HEADERS 64 /*!< Max amount of SIP headers to read */
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#define SIP_MAX_LINES 64 /*!< Max amount of lines in SIP attachment (like SDP) */
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#define SIP_MAX_PACKET 4096 /*!< Also from RFC 3261 (2543), should sub headers tho */
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static const char desc[] = "Session Initiation Protocol (SIP)";
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static const char channeltype[] = "SIP";
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static const char config[] = "sip.conf";
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static const char notify_config[] = "sip_notify.conf";
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static int usecnt = 0;
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#define RTP 1
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#define NO_RTP 0
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@ -338,7 +339,6 @@ static const struct cfsip_options {
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/*! \brief SIP Extensions we support */
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#define SUPPORTED_EXTENSIONS "replaces"
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#define SIP_MAX_PACKET 4096 /*!< Also from RFC 3261 (2543), should sub headers tho */
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/* Default values, set and reset in reload_config before reading configuration */
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/* These are default values in the source. There are other recommended values in the
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@ -380,11 +380,12 @@ static char default_vmexten[AST_MAX_EXTENSION];
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static char default_musicclass[MAX_MUSICCLASS]; /*!< Global music on hold class */
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/* Global settings only apply to the channel */
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static int global_rtautoclear = 120;
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static int global_notifyringing; /*!< Send notifications on ringing */
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static int srvlookup; /*!< SRV Lookup on or off. Default is off, RFC behavior is on */
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static int pedanticsipchecking; /*!< Extra checking ? Default off */
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static int autocreatepeer; /*!< Auto creation of peers at registration? Default off. */
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static int relaxdtmf; /*!< Relax DTMF */
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static int global_relaxdtmf; /*!< Relax DTMF */
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static int global_rtptimeout; /*!< Time out call if no RTP */
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static int global_rtpholdtimeout;
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static int global_rtpkeepalive; /*!< Send RTP keepalives */
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@ -401,6 +402,7 @@ static char global_realm[MAXHOSTNAMELEN]; /*!< Default realm */
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static char regcontext[AST_MAX_CONTEXT]; /*!< Context for auto-extensions */
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static char global_useragent[AST_MAX_EXTENSION]; /*!< Useragent for the SIP channel */
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static int allow_external_domains; /*!< Accept calls to external SIP domains? */
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static int global_callevents; /*!< Whether we send manager events or not */
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/*! \brief Codecs that we support by default: */
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static int global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263;
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@ -417,8 +419,6 @@ static int regobjs = 0; /*!< Registry objects */
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static struct ast_flags global_flags = {0}; /*!< global SIP_ flags */
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static struct ast_flags global_flags_page2 = {0}; /*!< more global SIP_ flags */
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static int usecnt =0;
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AST_MUTEX_DEFINE_STATIC(usecnt_lock);
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AST_MUTEX_DEFINE_STATIC(rand_lock); /*!< Lock for thread-safe random generator */
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@ -432,29 +432,23 @@ AST_MUTEX_DEFINE_STATIC(netlock);
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AST_MUTEX_DEFINE_STATIC(monlock);
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AST_MUTEX_DEFINE_STATIC(sip_reload_lock);
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/*! \brief This is the thread for the monitor which checks for input on the channels
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which are not currently in use. */
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static pthread_t monitor_thread = AST_PTHREADT_NULL;
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static int restart_monitor(void);
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static struct in_addr __ourip;
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static struct sockaddr_in outboundproxyip;
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static int ourport;
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static struct sockaddr_in debugaddr;
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static int sip_reloading = 0; /*!< Flag for avoiding multiple reloads at the same time */
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static enum channelreloadreason sip_reloadreason; /*!< Reason for last reload/load of configuration */
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static struct sched_context *sched;
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static struct io_context *io;
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#define DEC_CALL_LIMIT 0
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#define INC_CALL_LIMIT 1
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static struct ast_codec_pref prefs;
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/*! \brief sip_request: The data grabbed from the UDP socket */
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struct sip_request {
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char *rlPart1; /*!< SIP Method Name or "SIP/2.0" protocol version */
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@ -521,6 +515,7 @@ struct sip_auth {
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struct sip_auth *next; /*!< Next auth structure in list */
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};
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/*--- Various flags for the flags field in the pvt structure */
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#define SIP_ALREADYGONE (1 << 0) /*!< Whether or not we've already been destroyed by our peer */
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#define SIP_NEEDDESTROY (1 << 1) /*!< if we need to be destroyed */
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#define SIP_NOVIDEO (1 << 2) /*!< Didn't get video in invite, don't offer */
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@ -573,7 +568,7 @@ struct sip_auth {
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/* Remote Party-ID Support */
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#define SIP_SENDRPID (1 << 30)
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/* Did this connection increment the counter of in-use calls? */
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#define SIP_INC_COUNT (1 << 31)
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#define SIP_INC_COUNT (1 << 31)
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#define SIP_FLAGS_TO_COPY \
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(SIP_PROMISCREDIR | SIP_TRUSTRPID | SIP_SENDRPID | SIP_DTMF | SIP_REINVITE | \
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@ -598,7 +593,6 @@ struct sip_auth {
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#define sipdebug_config ast_test_flag(&global_flags_page2, SIP_PAGE2_DEBUG_CONFIG)
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#define sipdebug_console ast_test_flag(&global_flags_page2, SIP_PAGE2_DEBUG_CONSOLE)
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static int global_rtautoclear = 120;
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/*! \brief sip_pvt: PVT structures are used for each SIP dialog, ie. a call, a registration, a subscribe */
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static struct sip_pvt {
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@ -816,9 +810,6 @@ struct sip_peer {
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int lastmsg;
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};
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AST_MUTEX_DEFINE_STATIC(sip_reload_lock);
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static int sip_reloading = 0;
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static enum channelreloadreason sip_reloadreason; /*!< Reason for last reload/load of configuration */
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/* States for outbound registrations (with register= lines in sip.conf */
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#define REG_STATE_UNREGISTERED 0 /*!< We are not registred */
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@ -863,6 +854,8 @@ struct sip_registry {
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char lastmsg[256]; /*!< Last Message sent/received */
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};
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/* --- Linked lists of various objects --------*/
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/*! \brief The user list: Users and friends */
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static struct ast_user_list {
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ASTOBJ_CONTAINER_COMPONENTS(struct sip_user);
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@ -879,23 +872,25 @@ static struct ast_register_list {
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int recheck;
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} regl;
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static int __sip_do_register(struct sip_registry *r);
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static int sipsock = -1;
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/*! \todo Move the sip_auth list to AST_LIST */
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static struct sip_auth *authl = NULL; /*!< Authentication list for realm authentication */
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/* --- Sockets and networking --------------*/
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static int sipsock = -1; /*!< Main socket for SIP network communication */
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static struct sockaddr_in bindaddr = { 0, }; /*!< The address we bind to */
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static struct sockaddr_in externip; /*!< External IP address if we are behind NAT */
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static char externhost[MAXHOSTNAMELEN]; /*!< External host name (possibly with dynamic DNS and DHCP */
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static time_t externexpire = 0; /*!< Expiration counter for re-resolving external host name in dynamic DNS */
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static int externrefresh = 10;
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static struct ast_ha *localaddr; /*!< List of local networks, on the same side of NAT as this Asterisk */
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static int callevents; /*!< Whether we send manager events or not */
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static struct in_addr __ourip;
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static struct sockaddr_in outboundproxyip;
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static int ourport;
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static struct sockaddr_in debugaddr;
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struct ast_config *notify_types; /*!< The list of manual NOTIFY types we know how to send */
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static struct sip_auth *authl = NULL; /*!< Authentication list for realm authentication */
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/*---------------------------- Forward declarations of functions in chan_sip.c */
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@ -953,6 +948,8 @@ static void copy_request(struct sip_request *dst,struct sip_request *src);
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static int transmit_response_reliable(struct sip_pvt *p, char *msg, struct sip_request *req, int fatal);
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static int transmit_register(struct sip_registry *r, int sipmethod, char *auth, char *authheader);
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static int sip_poke_peer(struct sip_peer *peer);
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static int __sip_do_register(struct sip_registry *r);
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static int restart_monitor(void);
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/*! \brief Definition of this channel for PBX channel registration */
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static const struct ast_channel_tech sip_tech = {
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@ -2842,7 +2839,7 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
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if (ast_test_flag(i, SIP_DTMF) == SIP_DTMF_INBAND) {
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i->vad = ast_dsp_new();
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ast_dsp_set_features(i->vad, DSP_FEATURE_DTMF_DETECT);
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if (relaxdtmf)
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if (global_relaxdtmf)
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ast_dsp_digitmode(i->vad, DSP_DIGITMODE_DTMF | DSP_DIGITMODE_RELAXDTMF);
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}
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if (i->rtp) {
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@ -3748,7 +3745,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
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if (sin.sin_addr.s_addr && !sendonly) {
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append_history(p, "Unhold", "%s", req->data);
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if (callevents && ast_test_flag(p, SIP_CALL_ONHOLD)) {
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if (global_callevents && ast_test_flag(p, SIP_CALL_ONHOLD)) {
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manager_event(EVENT_FLAG_CALL, "Unhold",
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"Channel: %s\r\n"
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"Uniqueid: %s\r\n",
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@ -3761,7 +3758,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
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/* No address for RTP, we're on hold */
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append_history(p, "Hold", "%s", req->data);
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if (callevents && !ast_test_flag(p, SIP_CALL_ONHOLD)) {
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if (global_callevents && !ast_test_flag(p, SIP_CALL_ONHOLD)) {
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manager_event(EVENT_FLAG_CALL, "Hold",
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"Channel: %s\r\n"
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"Uniqueid: %s\r\n",
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@ -8302,7 +8299,7 @@ static int sip_show_settings(int fd, int argc, char *argv[])
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ast_cli(fd, " Caller ID: %s\n", default_callerid);
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ast_cli(fd, " From: Domain: %s\n", default_fromdomain);
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ast_cli(fd, " Record SIP history: %s\n", recordhistory ? "On" : "Off");
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ast_cli(fd, " Call Events: %s\n", callevents ? "On" : "Off");
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ast_cli(fd, " Call Events: %s\n", global_callevents ? "On" : "Off");
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ast_cli(fd, " IP ToS: 0x%x\n", global_tos);
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#ifdef OSP_SUPPORT
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ast_cli(fd, " OSP Support: Yes\n");
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@ -8319,7 +8316,7 @@ static int sip_show_settings(int fd, int argc, char *argv[])
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ast_cli(fd, " Codecs: ");
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print_codec_to_cli(fd, &prefs);
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ast_cli(fd, "\n");
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ast_cli(fd, " Relax DTMF: %s\n", relaxdtmf ? "Yes" : "No");
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ast_cli(fd, " Relax DTMF: %s\n", global_relaxdtmf ? "Yes" : "No");
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ast_cli(fd, " Compact SIP headers: %s\n", compactheaders ? "Yes" : "No");
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ast_cli(fd, " RTP Timeout: %d %s\n", global_rtptimeout, global_rtptimeout ? "" : "(Disabled)" );
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ast_cli(fd, " RTP Hold Timeout: %d %s\n", global_rtpholdtimeout, global_rtpholdtimeout ? "" : "(Disabled)");
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@ -12431,8 +12428,8 @@ static int reload_config(enum channelreloadreason reason)
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ast_clear_flag(&global_flags_page2, SIP_PAGE2_DEBUG_CONFIG);
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/* Misc settings for the channel */
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relaxdtmf = 0;
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callevents = 0;
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global_relaxdtmf = 0;
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global_callevents = 0;
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/* Read the [general] config section of sip.conf (or from realtime config) */
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for (v = ast_variable_browse(cfg, "general"); v; v = v->next) {
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@ -12463,7 +12460,7 @@ static int reload_config(enum channelreloadreason reason)
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} else if (!strcasecmp(v->name, "usereqphone")) {
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ast_set2_flag((&global_flags), ast_true(v->value), SIP_USEREQPHONE);
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} else if (!strcasecmp(v->name, "relaxdtmf")) {
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relaxdtmf = ast_true(v->value);
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global_relaxdtmf = ast_true(v->value);
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} else if (!strcasecmp(v->name, "checkmwi")) {
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if ((sscanf(v->value, "%d", &global_mwitime) != 1) || (global_mwitime < 0)) {
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ast_log(LOG_WARNING, "'%s' is not a valid MWI time setting at line %d. Using default (10).\n", v->value, v->lineno);
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@ -12620,7 +12617,7 @@ static int reload_config(enum channelreloadreason reason)
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default_qualify = 0;
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}
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} else if (!strcasecmp(v->name, "callevents")) {
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callevents = ast_true(v->value);
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global_callevents = ast_true(v->value);
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}
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}
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